Moved: The folder classes has been renamed to base

Updated: ultrastardx.dpr has been changed accordingly 

git-svn-id: svn://svn.code.sf.net/p/ultrastardx/svn/trunk@1339 b956fd51-792f-4845-bead-9b4dfca2ff2c

1 files changed, 0 insertions, 458 deletions

diff --git a/src/classes/UAudioConverter.pas b/src/classes/UAudioConverter.pas
deleted file mode 100644
index 5647f27b..00000000
--- a/src/classes/UAudioConverter.pas
+++ /dev/null
@@ -1,458 +0,0 @@
-unit UAudioConverter;
-
-interface
-
-{$IFDEF FPC}
-  {$MODE Delphi}
-{$ENDIF}
-
-{$I switches.inc}
-
-uses
-  UMusic,
-  ULog,
-  ctypes,
-  {$IFDEF UseSRCResample}
-  samplerate,
-  {$ENDIF}
-  {$IFDEF UseFFmpegResample}
-  avcodec,
-  {$ENDIF}
-  UMediaCore_SDL,
-  sdl,
-  SysUtils,
-  Math;
-
-type
-  {*
-   * Notes:
-   *  - 44.1kHz to 48kHz conversion or vice versa is not supported
-   *    by SDL 1.2 (will be introduced in 1.3).
-   *    No conversion takes place in this cases.
-   *    This is because SDL just converts differences in powers of 2.
-   *    So the result might not be that accurate.
-   *    This IS audible (voice to high/low) and it needs good synchronization
-   *    with the video or the lyrics timer.
-   *  - float<->int16 conversion is not supported (will be part of 1.3) and
-   *    SDL (<1.3) is not capable of handling floats at all.
-   *  -> Using FFmpeg or libsamplerate for resampling is preferred.
-   *     Use SDL for channel and format conversion only.
-   *}
-  TAudioConverter_SDL = class(TAudioConverter)
-    private
-      cvt: TSDL_AudioCVT;
-    public
-      function Init(SrcFormatInfo: TAudioFormatInfo; DstFormatInfo: TAudioFormatInfo): boolean; override;
-      destructor Destroy(); override;
-
-      function Convert(InputBuffer: PChar; OutputBuffer: PChar; var InputSize: integer): integer; override;
-      function GetOutputBufferSize(InputSize: integer): integer; override;
-      function GetRatio(): double; override;
-  end;
-
-  {$IFDEF UseFFmpegResample}
-  // Note: FFmpeg seems to be using "kaiser windowed sinc" for resampling, so
-  // the quality should be good.
-  TAudioConverter_FFmpeg = class(TAudioConverter)
-    private
-      // TODO: use SDL for multi-channel->stereo and format conversion
-      ResampleContext: PReSampleContext;
-      Ratio: double;
-    public
-      function Init(SrcFormatInfo: TAudioFormatInfo; DstFormatInfo: TAudioFormatInfo): boolean; override;
-      destructor Destroy(); override;
-
-      function Convert(InputBuffer: PChar; OutputBuffer: PChar; var InputSize: integer): integer; override;
-      function GetOutputBufferSize(InputSize: integer): integer; override;
-      function GetRatio(): double; override;
-  end;
-  {$ENDIF}
-
-  {$IFDEF UseSRCResample}
-  TAudioConverter_SRC = class(TAudioConverter)
-    private
-      ConverterState: PSRC_STATE;
-      ConversionData: SRC_DATA;
-      FormatConverter: TAudioConverter;
-    public
-      function Init(SrcFormatInfo: TAudioFormatInfo; DstFormatInfo: TAudioFormatInfo): boolean; override;
-      destructor Destroy(); override;
-
-      function Convert(InputBuffer: PChar; OutputBuffer: PChar; var InputSize: integer): integer; override;
-      function GetOutputBufferSize(InputSize: integer): integer; override;
-      function GetRatio(): double; override;
-  end;
-
-  // Note: SRC (=libsamplerate) provides several converters with different quality
-  // speed trade-offs. The SINC-types are slow but offer best quality.
-  // The SRC_SINC_* converters are too slow for realtime conversion,
-  // (SRC_SINC_FASTEST is approx. ten times slower than SRC_LINEAR) resulting
-  // in audible clicks and pops.
-  // SRC_LINEAR is very fast and should have a better quality than SRC_ZERO_ORDER_HOLD
-  // because it interpolates the samples. Normal "non-audiophile" users should not
-  // be able to hear a difference between the SINC_* ones and LINEAR. Especially
-  // if people sing along with the song.
-  // But FFmpeg might offer a better quality/speed ratio than SRC_LINEAR.
-  const
-    SRC_CONVERTER_TYPE = SRC_LINEAR; 
-  {$ENDIF}
-
-implementation
-
-function TAudioConverter_SDL.Init(srcFormatInfo: TAudioFormatInfo; dstFormatInfo: TAudioFormatInfo): boolean;
-var
-  srcFormat: UInt16;
-  dstFormat: UInt16;
-begin
-  inherited Init(SrcFormatInfo, DstFormatInfo);
-
-  Result := false;
-
-  if (not ConvertAudioFormatToSDL(srcFormatInfo.Format, srcFormat) or
-      not ConvertAudioFormatToSDL(dstFormatInfo.Format, dstFormat)) then
-  begin
-    Log.LogError('Audio-format not supported by SDL', 'TSoftMixerPlaybackStream.InitFormatConversion');
-    Exit;
-  end;
-
-  if (SDL_BuildAudioCVT(@cvt,
-    srcFormat, srcFormatInfo.Channels, Round(srcFormatInfo.SampleRate),
-    dstFormat, dstFormatInfo.Channels, Round(dstFormatInfo.SampleRate)) = -1) then
-  begin
-    Log.LogError(SDL_GetError(), 'TSoftMixerPlaybackStream.InitFormatConversion');
-    Exit;
-  end;
-
-  Result := true;
-end;
-
-destructor TAudioConverter_SDL.Destroy();
-begin
-  // nothing to be done here
-  inherited;
-end;
-
-(*
- * Returns the size of the output buffer. This might be bigger than the actual
- * size of resampled audio data.
- *)
-function TAudioConverter_SDL.GetOutputBufferSize(InputSize: integer): integer;
-begin
-  // Note: len_ratio must not be used here. Even if the len_ratio is 1.0, len_mult might be 2.
-  // Example: 44.1kHz/mono to 22.05kHz/stereo -> len_ratio=1, len_mult=2
-  Result := InputSize * cvt.len_mult;
-end;
-
-function TAudioConverter_SDL.GetRatio(): double;
-begin
-  Result := cvt.len_ratio;
-end;
-
-function TAudioConverter_SDL.Convert(InputBuffer: PChar; OutputBuffer: PChar; var InputSize: integer): integer;
-begin
-  Result := -1;
-
-  if (InputSize <= 0) then
-  begin
-    // avoid div-by-zero problems
-    if (InputSize = 0) then
-      Result := 0;
-    Exit;
-  end;
-
-  // OutputBuffer is always bigger than or equal to InputBuffer
-  Move(InputBuffer[0], OutputBuffer[0], InputSize);
-  cvt.buf := PUint8(OutputBuffer);
-  cvt.len := InputSize;
-  if (SDL_ConvertAudio(@cvt) = -1) then
-    Exit;
-
-  Result := cvt.len_cvt;
-end;
-
-
-{$IFDEF UseFFmpegResample}
-
-function TAudioConverter_FFmpeg.Init(SrcFormatInfo: TAudioFormatInfo; DstFormatInfo: TAudioFormatInfo): boolean;
-begin
-  inherited Init(SrcFormatInfo, DstFormatInfo);
-
-  Result := false;
-
-  // Note: ffmpeg does not support resampling for more than 2 input channels
-
-  if (srcFormatInfo.Format <> asfS16) then
-  begin
-    Log.LogError('Unsupported format', 'TAudioConverter_FFmpeg.Init');
-    Exit;
-  end;
-
-  // TODO: use SDL here
-  if (srcFormatInfo.Format <> dstFormatInfo.Format) then
-  begin
-    Log.LogError('Incompatible formats', 'TAudioConverter_FFmpeg.Init');
-    Exit;
-  end;
-
-  ResampleContext := audio_resample_init(
-      dstFormatInfo.Channels, srcFormatInfo.Channels,
-      Round(dstFormatInfo.SampleRate), Round(srcFormatInfo.SampleRate));
-  if (ResampleContext = nil) then
-  begin
-    Log.LogError('audio_resample_init() failed', 'TAudioConverter_FFmpeg.Init');
-    Exit;
-  end;
-
-  // calculate ratio
-  Ratio := (dstFormatInfo.Channels / srcFormatInfo.Channels) *
-           (dstFormatInfo.SampleRate / srcFormatInfo.SampleRate);
-
-  Result := true;
-end;
-
-destructor TAudioConverter_FFmpeg.Destroy();
-begin
-  if (ResampleContext <> nil) then
-    audio_resample_close(ResampleContext);
-  inherited;
-end;
-
-function TAudioConverter_FFmpeg.Convert(InputBuffer: PChar; OutputBuffer: PChar; var InputSize: integer): integer;
-var
-  InputSampleCount: integer;
-  OutputSampleCount: integer;
-begin
-  Result := -1;
-
-  if (InputSize <= 0) then
-  begin
-    // avoid div-by-zero in audio_resample()
-    if (InputSize = 0) then
-      Result := 0;
-    Exit;
-  end;
-
-  InputSampleCount := InputSize div SrcFormatInfo.FrameSize;
-  OutputSampleCount := audio_resample(
-      ResampleContext, PSmallInt(OutputBuffer), PSmallInt(InputBuffer),
-      InputSampleCount);
-  if (OutputSampleCount = -1) then
-  begin
-    Log.LogError('audio_resample() failed', 'TAudioConverter_FFmpeg.Convert');
-    Exit;
-  end;
-  Result := OutputSampleCount * DstFormatInfo.FrameSize;
-end;
-
-function TAudioConverter_FFmpeg.GetOutputBufferSize(InputSize: integer): integer;
-begin
-  Result := Ceil(InputSize * GetRatio());
-end;
-
-function TAudioConverter_FFmpeg.GetRatio(): double;
-begin
-  Result := Ratio;
-end;
-
-{$ENDIF}
-
-
-{$IFDEF UseSRCResample}
-
-function TAudioConverter_SRC.Init(SrcFormatInfo: TAudioFormatInfo; DstFormatInfo: TAudioFormatInfo): boolean;
-var
-  error: integer;
-  TempSrcFormatInfo: TAudioFormatInfo;
-  TempDstFormatInfo: TAudioFormatInfo;
-begin
-  inherited Init(SrcFormatInfo, DstFormatInfo);
-
-  Result := false;
-
-  FormatConverter := nil;
-
-  // SRC does not handle channel or format conversion
-  if ((SrcFormatInfo.Channels <> DstFormatInfo.Channels) or
-      not (SrcFormatInfo.Format in [asfS16, asfFloat])) then
-  begin
-    // SDL can not convert to float, so we have to convert to SInt16 first
-    TempSrcFormatInfo := TAudioFormatInfo.Create(
-        SrcFormatInfo.Channels, SrcFormatInfo.SampleRate, SrcFormatInfo.Format);
-    TempDstFormatInfo := TAudioFormatInfo.Create(
-        DstFormatInfo.Channels, SrcFormatInfo.SampleRate, asfS16);
-
-    // init format/channel conversion
-    FormatConverter := TAudioConverter_SDL.Create();
-    if (not FormatConverter.Init(TempSrcFormatInfo, TempDstFormatInfo)) then
-    begin
-      Log.LogError('Unsupported input format', 'TAudioConverter_SRC.Init');
-      FormatConverter.Free;
-      // exit after the format-info is freed
-    end;
-
-    // this info was copied so we do not need it anymore 
-    TempSrcFormatInfo.Free;
-    TempDstFormatInfo.Free;
-
-    // leave if the format is not supported
-    if (not assigned(FormatConverter)) then
-      Exit;
-
-    // adjust our copy of the input audio-format for SRC conversion
-    Self.SrcFormatInfo.Channels := DstFormatInfo.Channels;
-    Self.SrcFormatInfo.Format := asfS16;
-  end;
-
-  if ((DstFormatInfo.Format <> asfS16) and
-      (DstFormatInfo.Format <> asfFloat)) then
-  begin
-    Log.LogError('Unsupported output format', 'TAudioConverter_SRC.Init');
-    Exit;
-  end;
-
-  ConversionData.src_ratio := DstFormatInfo.SampleRate / SrcFormatInfo.SampleRate;
-  if (src_is_valid_ratio(ConversionData.src_ratio) = 0) then
-  begin
-    Log.LogError('Invalid samplerate ratio', 'TAudioConverter_SRC.Init');
-    Exit;
-  end;
-
-  ConverterState := src_new(SRC_CONVERTER_TYPE, DstFormatInfo.Channels, @error);
-  if (ConverterState = nil) then
-  begin
-    Log.LogError('src_new() failed: ' + src_strerror(error), 'TAudioConverter_SRC.Init');
-    Exit;
-  end;
-
-  Result := true;
-end;
-
-destructor TAudioConverter_SRC.Destroy();
-begin
-  if (ConverterState <> nil) then
-    src_delete(ConverterState);
-  FormatConverter.Free;
-  inherited;
-end;
-
-function TAudioConverter_SRC.Convert(InputBuffer: PChar; OutputBuffer: PChar; var InputSize: integer): integer;
-var
-  FloatInputBuffer: PSingle;
-  FloatOutputBuffer: PSingle;
-  TempBuffer: PChar;
-  TempSize: integer;
-  NumSamples: integer;
-  OutputSize: integer;
-  error: integer;
-begin
-  Result := -1;
-
-  TempBuffer := nil;
-
-  // format conversion with external converter (to correct number of channels and format)
-  if (assigned(FormatConverter)) then
-  begin
-    TempSize := FormatConverter.GetOutputBufferSize(InputSize);
-    GetMem(TempBuffer, TempSize);
-    InputSize := FormatConverter.Convert(InputBuffer, TempBuffer, InputSize);
-    InputBuffer := TempBuffer;
-  end;
-
-  if (InputSize <= 0) then
-  begin
-    // avoid div-by-zero problems
-    if (InputSize = 0) then
-      Result := 0;
-    if (TempBuffer <> nil) then
-      FreeMem(TempBuffer);
-    Exit;
-  end;
-
-  if (SrcFormatInfo.Format = asfFloat) then
-  begin
-    FloatInputBuffer := PSingle(InputBuffer);
-  end else begin
-    NumSamples := InputSize div AudioSampleSize[SrcFormatInfo.Format];
-    GetMem(FloatInputBuffer, NumSamples * SizeOf(Single));
-    src_short_to_float_array(PCshort(InputBuffer), PCfloat(FloatInputBuffer), NumSamples);
-  end;
-
-  // calculate approx. output size
-  OutputSize := Ceil(InputSize * ConversionData.src_ratio);
-
-  if (DstFormatInfo.Format = asfFloat) then
-  begin
-    FloatOutputBuffer := PSingle(OutputBuffer);
-  end else begin
-    NumSamples := OutputSize div AudioSampleSize[DstFormatInfo.Format];
-    GetMem(FloatOutputBuffer, NumSamples * SizeOf(Single));
-  end;
-
-  with ConversionData do
-  begin
-    data_in := PCFloat(FloatInputBuffer);
-    input_frames := InputSize div SrcFormatInfo.FrameSize;
-    data_out := PCFloat(FloatOutputBuffer);
-    output_frames := OutputSize div DstFormatInfo.FrameSize;
-    // TODO: set this to 1 at end of file-playback
-    end_of_input := 0;
-  end;
-
-  error := src_process(ConverterState, @ConversionData);
-  if (error <> 0) then
-  begin
-    Log.LogError(src_strerror(error), 'TAudioConverter_SRC.Convert');
-    if (SrcFormatInfo.Format <> asfFloat) then
-      FreeMem(FloatInputBuffer);
-    if (DstFormatInfo.Format <> asfFloat) then
-      FreeMem(FloatOutputBuffer);
-    if (TempBuffer <> nil) then
-      FreeMem(TempBuffer);
-    Exit;
-  end;
-
-  if (SrcFormatInfo.Format <> asfFloat) then
-    FreeMem(FloatInputBuffer);
-
-  if (DstFormatInfo.Format <> asfFloat) then
-  begin
-    NumSamples := ConversionData.output_frames_gen * DstFormatInfo.Channels;
-    src_float_to_short_array(PCfloat(FloatOutputBuffer), PCshort(OutputBuffer), NumSamples);
-    FreeMem(FloatOutputBuffer);
-  end;
-
-  // free format conversion buffer if used
-  if (TempBuffer <> nil) then
-    FreeMem(TempBuffer);
-
-  if (assigned(FormatConverter)) then
-    InputSize := ConversionData.input_frames_used * FormatConverter.SrcFormatInfo.FrameSize
-  else
-    InputSize := ConversionData.input_frames_used * SrcFormatInfo.FrameSize;
-
-  // set result to output size according to SRC
-  Result := ConversionData.output_frames_gen * DstFormatInfo.FrameSize;
-end;
-
-function TAudioConverter_SRC.GetOutputBufferSize(InputSize: integer): integer;
-begin
-  Result := Ceil(InputSize * GetRatio());
-end;
-
-function TAudioConverter_SRC.GetRatio(): double;
-begin
-  // if we need additional channel/format conversion, use this ratio
-  if (assigned(FormatConverter)) then
-    Result := FormatConverter.GetRatio()
-  else
-    Result := 1.0;
-
-  // now the SRC ratio (Note: the format might change from SInt16 to float)
-  Result := Result *
-            ConversionData.src_ratio *
-            (DstFormatInfo.FrameSize / SrcFormatInfo.FrameSize);
-end;
-
-{$ENDIF}
-
-end.
\ No newline at end of file